The Voice over Internet Protocol (VoIP) is a protocol that allows a user to make telephone calls using an Internet connection rather than a traditional analog telephone connection. With VoIP, the caller's voice signal is converted from an analog signal into a digital signal carried by IP packets that travel over the Internet. The digital signal is then converted back into a voice analog signal at the other end so that the caller can speak with a called party. VoIP uses the Internet as the transmission medium for telephone calls by sending voice data in packets using IP rather than by traditional circuit transmissions of the Public Switched Telephone Network (PSTN).
The Session Initiation Protocol (SIP) is a protocol used in VOIP network environments. SIP is a signaling and call setup protocol for IP-based communications. SIP has also been used to enable networks to implement many call processing features such as dialing sounds, causing a phone to ring and busy signals that provide the illusion of a normal telephone operation. Unfortunately, the use of SIP to provide these and other calling features have required intelligent call endpoints that support the use of SIP to support the same features. It would be desirable to use SIP to manage the creation and treatment of communication streams in a VOIP environment from a centralized location in the communication network so that features are not required to be supported at both call endpoints.